Philipp Hancke
Philipp Hancke
I was looking at https://webrtc.github.io/samples/src/content/peerconnection/audio/ which in Chrome 96+ shows audio/RED with a sdpFmtpLine 111/111. This pulls from RTCRtpSender.getCapabilities("audio").codecs ``` [ { "channels": 2, "clockRate": 48000, "mimeType": "audio/opus", "sdpFmtpLine": "minptime=10;useinbandfec=1"...
https://datatracker.ietf.org/meeting/111/materials/slides-111-rtcweb-aligning-jsep-and-bundle-00 seems to be the latest on that. tl;dr is max-bundle is deprecated and must-bundle needs to be added. It also seems we have implementation issues, neither Chrome nor Firefox...
Writing a test about rejecting SDES I found that browsers differ in the type of error they reject the following with: ``` const sdp = `v=0 o=- 0 3 IN...
using broadcast channel like for https://webrtc.github.io/samples/src/content/peerconnection/channel/ Fixes #1325
note: this will work in M105+ once https://chromiumdash.appspot.com/commit/1fe14f2752a35932bea6d6ccff7433a6716c6391 is available
https://developer.chrome.com/docs/privacy-sandbox/permissions-policy/ has some nice things which would be good to show Also covers #1162
### Issue description From libwebrtc upstream: https://bugs.chromium.org/p/webrtc/issues/detail?id=14272 mediasoup has the similar (but not exactly the same, probably an old version) code [here](https://github.com/versatica/mediasoup/blame/v3/worker/deps/libwebrtc/libwebrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc#L394), the [fix](https://webrtc-review.googlesource.com/c/src/+/269002/2/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.cc) should still apply
fixes #764,#1017
https://api.npmjs.org/versions/webrtc-adapter/last-week shows the download distribution. Looks like a considerable amount of 6.x usage
based on enumerateDevices. This allows apps to somewhat use getUserMedia already to use selectAudioOutput (but will not show any prompt). Note that this does not (yet) modify enumerateDevices output.